[Serusers] Modifying "From" header field of an INVITE message

scm-j at nuntius.com scm-j at nuntius.com
Mon Feb 7 05:00:44 CET 2005


Thanks Girish, Greg, John and Andres. I made the change to the Asterisk
dialplan and the "From" field is now appropriately presented to SER.
Andres, I could not use re-invite as my application requires Asterisk to
bridge the RTP media path. Thanks a lot!!!

Another problem I face is when I call the toll-free number (that points
to my SER) from a phone with the caller ID blocked, although I receive
the ANI (from Level-3) and is passed onto Asterisk (also shows up in
$CALLERID), when Asterisk makes the outbound call (Dial command), it
changes the "From" field to "Unknown". However, if I make the same call
with the callerID unblocked, the ANI is present in the "From" field. 

Look like there is some kind of privacy field in the SIP (or SDP) header
that triggers Asterisk to block the callerID on calls originating from
blocked numbers. The problem I face due to this, is that I am unable to
terminate calls to other toll-free numbers, as they require the ANI. Any
clues?

regards,
SCM

-----Original Message-----
From: Andrés Parra L. [mailto:apl_1980b at yahoo.com] 
Sent: Sunday, February 06, 2005 11:26 AM
To: serusers at lists.iptel.org
Subject: Re: [Serusers] Modifying "From" header field of an INVITE
message

Don't hesitate anymore, use asterisk to do so with the
SetCallerID command in extensions.conf. You should
forward the call through Asterisk from SER. If you
don't want to have all the media pass through you, and
i beleive you don't, use canreinvite=yes in sip.conf,
you should fight a little bit with the codecs with
allow and disallow but it should be fine. The next
links could help you with the reinvitation issue:
http://www.voip-info.org/wiki-Asterisk+SIP+media+path
http://www.voip-info.org/wiki-Asterisk+sip+canreinvite


Hope it helps.

Andres


		
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